June 2014 - VoiceOps - lists.voiceops.org

Maintenance for Performance Technologies SEGway
by scott＠socket.net May 17, 2017

May 17, 2017

We have been using a set of PTI SEGway x201s with point code emulation for a number of years now. We also have an unlicensed spare in stock. The maintenance contract renewals are due and the two quotes I've received are quite high. I know of at least one other company that has chosen to drop the vendor maintenance and take care of that in-house. I would like to hear from anyone that is currently using SEGways and find out what they chose to do for maintenance. I would also be very interested to hear of any bad experiences from chosing the in-house option. Thanks. Scott Stratman Network Facilities Manager SOCKET 573-999-4376 scott at socket.net 1-800-SOCKET-3 www.socket.net

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Monitoring tools
by jra＠baylink.com June 30, 2014

June 30, 2014

The Telephony plugin to Wireshark is pretty neat, but moreso for diagnosis than monitoring. Is there -- this would probably be a PCAP client app -- something that just monitors a port for SIP/STP flows, and shows a call progress thermometer, with a flow number, destination or source DN, direction, and some color coding to show you that the call is proceeding as expected or is anomalous? Basically, something that launches a small state machine for each new flow it seems, and displays the status of the state machine as a row in a table? Clearly, this would get out of hand for hundreds or thousands of calls, but dozens is a pretty big market... Cheers, -- jra -- Jay R. Ashworth Baylink jra at baylink.com Designer The Things I Think RFC 2100 Ashworth & Associates http://www.bcp38.info 2000 Land Rover DII St Petersburg FL USA BCP38: Ask For It By Name! +1 727 647 1274

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PDD: What's acceptable?
by david＠ringfree.biz June 26, 2014

June 26, 2014

I'm curious how everyone handles this.. Do you run reports and open tickets proactively with vendors that have high PDD? Do you just wait for clients to complain and than address specific numbers? I've recently been thinking about running reports and everything over 10 seconds gets reported to the vendor in question. But I don't want them to hate me. Thanks David -- -- www.ringfree.biz 828-575-0030

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Solving port blocking issue @ softphone
by feby.francis＠crosstel.com June 26, 2014

June 26, 2014

Experts, I have a customer who travels a lot and uses our softphone offering, I need a solution to overcome the situations when the port 5060 is blocked. Is there any work around to solve the port blocking problems? We use standard 5060 on the SBC and my switch vendor is Broadsoft. Thanks for all the help and advices, Feby Francis

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Call established before audio path
by briant＠socket.net June 26, 2014

June 26, 2014

We have customers who are reporting that when people call into them that the beginning of the greeting is not being heard. It appears that the phones think the call is ready before there is a complete audio path. We see this the most on calls that come in from the PSTN. Long distance or toll free calls are the worst. We have a Broadworks system that uses Lucent 5010 (Telica) switches for the connections to the PSTN. The symptom seems to be very similar for all of our PSTN connections, to multiple carriers. I'm just curious if anyone has any experience with this problem, or if anyone has any suggestions to get to the bottom of the issue. Any assistance or inspiration would be greatly appreciated. Thank you, Brian Taylor Socket Telecom, LLC brian at socket.net

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Monitoring tools
by jra＠baylink.com June 25, 2014

June 25, 2014

[ inadvertantly sent to NANOG. I hate RFC 2919 non-compliant mailers ] ----- Original Message ----- > From: "Joseph Jackson" <jjackson at aninetworks.net> > Check out Homer @ http://sipcapture.org we love it. There is also a > commercial version that has more features / support. Well, I don't know if Homer hsa the realtime viz tool I'm looking for, but if it lives up to the attitude of it's website and prezi, then it's probably a good thing that I know about it anyway, thanks. Wonder if anyone's working on Sonus support. Cheers, -- jra -- Jay R. Ashworth Baylink jra at baylink.com Designer The Things I Think RFC 2100 Ashworth & Associates http://www.bcp38.info 2000 Land Rover DII St Petersburg FL USA BCP38: Ask For It By Name! +1 727 647 1274

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HIPAA compliance
by ujjval＠simplesignal.com June 25, 2014

June 25, 2014

Hi Team Voiceops: What is needed for an ITSP to be HIPPA complaint? Sounds like a really dark grey area! Interesting Article here: http://www.linkedin.com/groups/Recommended-HIPAA-Compliant-VoIP-Services-22… Ujjval Karihaloo

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Odd network problem with SIP
by jra＠baylink.com June 24, 2014

June 24, 2014

----- Original Message ----- > From: "Frank Bulk" <frnkblk at iname.com> > Except, by definition, jitter is ping-response (latency) deviation. In > calculus terms, jitter is the derivative (change in rate) of latency. I was assuming that jitter had to do with the sliding of packets around WRT strict isochronous positioning -- specifically, packets arriving out of sequence, because they are so late. That's not what jitter means in VoIP? :-) Cheers, -- jra -- Jay R. Ashworth Baylink jra at baylink.com Designer The Things I Think RFC 2100 Ashworth & Associates http://www.bcp38.info 2000 Land Rover DII St Petersburg FL USA BCP38: Ask For It By Name! +1 727 647 1274

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Odd network problem with SIP
by jra＠baylink.com June 21, 2014

June 21, 2014

I have a client using the Spitfire dialler on Win7 to dial via SIP to a carrier called VoxTelecom; I infer they're a virtual carrier because the SIP goes to their Sonus SBC at a single IP, but the media sessions go all over creation. The client has been having completion trouble, and the dialler folks said "crappy circuit; too much jitter and packet loss", tested from the dialler using PingPlotter. I don't think it was jitter, I think it was ping response deviation; PingPlotter doesn't appear to actually test jitter. But to tick the boxes, I had road runner out; they replaced a ubee D2 modem with their New Hawtness Arris D3 modem; no change. All his analog measurements were pristine, he told me. So next step, check the router. Log in to the Fortigate 40D; control panel won't paint properly. This *may* have been an IE10 compatibility mode red-herring, but I temporarily swapped it for a 20C, which I could talk to. Set up the inbound DNATs for udp/5060 SIP and upd/49152-49252 RTP. Ran some calls. They're not seeing my ACK after they send me an SDP. They get the invite, but not the ACK. I send RTP, but they don't bother cause they think the call's not set up yet. Wireshark on the dialler... that ACK packet *has a bad IP header checksum*. Not the earlier packets; just the ACK. Huh? So, assume it's the OS, somehow; reboot. 65 Windows updates and an hour later... Set everything back up, and run more test calls. This time, the Invite has a bad IP header checksum. Look at clock, 6pm EDT. Give up, go home. I'm going to go back to the original router Monday morning, and check the compatibility mode theory, but has anyone ever seen "just certain sent packets show up with a bad IP header checksum, as monitored on-machine"? (I know that the original 40D router might well *have* a problem, and that the checksum thing is orthogonal to the original problem -- since older pcaps show clean setups -- but at least I can confirm I have the NATs configured right.) Cheers, -- jra -- Jay R. Ashworth Baylink jra at baylink.com Designer The Things I Think RFC 2100 Ashworth & Associates http://www.bcp38.info 2000 Land Rover DII St Petersburg FL USA BCP38: Ask For It By Name! +1 727 647 1274

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Asterisk Inconsistently sending RTP for deterministic set of orig DIDs
by beckman＠angryox.com June 19, 2014

June 19, 2014

I'm dealing with a strange situation and I'm hoping there might be someone who can see an easy answer. 1. When a call comes into Asterisk, we answer the call and send a SIP 200 OK. Then we play an audio clip, then bridge the call to a Dial() with the 'r' option to play ringing to the origination side of the call. This works with all of Carrier B's DIDs and most of Carrier A's DIDs. The RTP streams start sending as soon as we answer() the inbound call. 2. For other DIDs, we answer() and send a 200 OK and do not play any audio but bridge the call directly to a Dial(). The audio is still passed to the caller and they hear ringing generated from Asterisk, not locally. 3. For the problem DIDs we are working on right now, they all look like #2 but no audio is passed to the caller, and the RTP stream is not sent (based on tcpdumps). However when the Dial()ed call leg answers, the RTP stream begins (inconsistently, but that's another issue). Because this issue is happening with just a small subset of DIDs on one specific carrier of ours, and that we have made NO changes to our Asterisk configuration or our AGI that handles calls, and that the same AGI handles all inbound calls the same way, I'm looking for any troubleshooting advice I can find. This started very suddenly after several years of no issues three days ago. My Asterisk server has enough inodes, very few open files (no where close to ulimit levels), and no indication that there are any problems. There are no limitations on inbound or outbound ports for RTP (no firewall rules restricting that traffic). Reinvites are not enabled. Beckman --------------------------------------------------------------------------- Peter Beckman Internet Guy beckman at angryox.com http://www.angryox.com/ ---------------------------------------------------------------------------

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